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- /* Audio Library for Teensy 3.X
- * Copyright (c) 2019, Paul Stoffregen, paul@pjrc.com
- *
- * Development of this audio library was funded by PJRC.COM, LLC by sales of
- * Teensy and Audio Adaptor boards. Please support PJRC's efforts to develop
- * open source software by purchasing Teensy or other PJRC products.
- *
- * Permission is hereby granted, free of charge, to any person obtaining a copy
- * of this software and associated documentation files (the "Software"), to deal
- * in the Software without restriction, including without limitation the rights
- * to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
- * copies of the Software, and to permit persons to whom the Software is
- * furnished to do so, subject to the following conditions:
- *
- * The above copyright notice, development funding notice, and this permission
- * notice shall be included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
- * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
- * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
- * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
- * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
- * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
- * THE SOFTWARE.
- */
- /*
- by Alexander Walch
- */
- #ifndef async_input_spdif3_h_
- #define async_input_spdif3_h_
- #include "./Resampler.h"
- #include "./Quantizer.h"
- #include <core/Arduino.h>
- #include <core/AudioStream.h>
- #include <core/DMAChannel.h>
- #include <arm_math.h>
-
- //#define DEBUG_SPDIF_IN //activates debug output
-
- class AsyncAudioInputSPDIF3 : public AudioStream
- {
- public:
- ///@param attenuation target attenuation [dB] of the anti-aliasing filter. Only used if AUDIO_SAMPLE_RATE_EXACT < input sample rate (input fs). The attenuation can't be reached if the needed filter length exceeds 2*MAX_FILTER_SAMPLES+1
- ///@param minHalfFilterLength If AUDIO_SAMPLE_RATE_EXACT >= input fs), the filter length of the resampling filter is 2*minHalfFilterLength+1. If AUDIO_SAMPLE_RATE_EXACT < input fs the filter is maybe longer to reach the desired attenuation
- ///@param maxHalfFilterLength Can be used to restrict the maximum filter length at the cost of a lower attenuation
- AsyncAudioInputSPDIF3(bool dither=false, bool noiseshaping=false,float attenuation=100, int32_t minHalfFilterLength=20, int32_t maxHalfFilterLength=80);
- ~AsyncAudioInputSPDIF3();
- void begin();
- virtual void update(void);
- double getBufferedTime() const;
- double getInputFrequency() const;
- static bool isLocked();
- double getTargetLantency() const;
- double getAttenuation() const;
- int32_t getHalfFilterLength() const;
- protected:
- static DMAChannel dma;
- static void isr(void);
- private:
- void resample(int16_t* data_left, int16_t* data_right, int32_t& block_offset);
- void monitorResampleBuffer();
- void configure();
- double getNewValidInputFrequ();
- void config_spdifIn();
-
- //accessed in isr ====
- static volatile int32_t buffer_offset;
- static int32_t resample_offset;
- static volatile uint32_t microsLast;
- //====================
-
- Resampler _resampler;
- Quantizer* quantizer[2];
- arm_biquad_cascade_df2T_instance_f32 _bufferLPFilter;
-
- volatile double _bufferedTime;
- volatile double _lastValidInputFrequ;
- double _inputFrequency=0.;
- double _targetLatencyS; //target latency [seconds]
- const double _blockDuration=AUDIO_BLOCK_SAMPLES/AUDIO_SAMPLE_RATE_EXACT; //[seconds]
- double _maxLatency=2.*_blockDuration;
-
- #ifdef DEBUG_SPDIF_IN
- static volatile bool bufferOverflow;
- #endif
- };
-
- #endif
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